1. Field of the Invention
The present invention relates generally to improvements in electronic filters, and more particularly, but not by way of limitation, to improvements in electronic filters suitable for use in sampled communication channels such as in a disc drive read channel.
2. Discussion
Historically, the term "filter" has generally referred to a device designed to exhibit a specified magnitude response to an input signal. Recent developments in sampled, pulse-based communication systems, however, have led to the realization that time-domain characteristics of a filter are as important a design consideration as the frequency-domain characteristics of the filter. Conventional frequency-domain analysis, though, is still predominant in the construction and operation of a filtering system, as the ability of a filter to reject random noise is best described in the frequency domain.
The importance of a filter's time-domain characteristics can be exemplified by what is commonly referred to as a "matched filter", which will be recognized as a filter having an optimized output signal-to-noise ratio when the input response of the filter is the mirror image of the received pulse. A matched filter is a true filter in every sense of the word, but the foregoing definition emphasizes the filter's time-domain characteristics and not its frequency response. Such a matched filter is "signal selective", rather than "frequency selective"; that is, a matched filter makes no attempt to preserve the shape of an input signal at the output of the filter, but rather attempts to optimize the detection of the input signal.
The time-domain characteristics of filters are a primary concern in the transmission of data through modern communications networks. Particularly, it will be recognized by those skilled in communication theory that an optimal linear filter for the binary transmission of information through a communication channel subject to dispersion and intersymbol interference (ISI) and immersed in white Gaussian noise (WGN) comprises a matched filter in series with a tapped delay line (the matched filter is matched to the received pulse, not the transmitted pulse). See, for example, the text R. D. Gitlin et al. entitled "DATA COMMUNICATION PRINCIPLES", Plenum Press, New York, 1992, pp. 491-508, incorporated herein by reference. In other words, the output of the optimal linear receiving filter is a weighted sum of the output of the matched filter which is delayed by different amounts of time nT (with n being an integer and T being a bit, or sampling period). The output of the linear receiving filter is then sampled at the bit rate T and this provides the best estimate of the transmitted bit stream.
The structure which produces a weighted sum of time-delayed versions of an input signal is known as a "transversal filter" or "transversal equalizer". It will be recognized that the optimum values for the tap weights can be determined adaptively by the transversal equalizer itself, and, once so optimized, the transversal equalizer operates as a time-domain filter to minimize the effects of ISI in the transmitted data stream. For more discussion, see A. Gersho, "ADAPTIVE EQUALIZATION OF HIGHLY DISPERSIVE CHANNELS FOR DATA TRANSMISSION", Bell System Technical Journal, January 1969, pp. 55-69, incorporated herein by reference.
In the past, it has not been generally feasible to realize an optimal analog transversal equalizer in a communication channel, due to a variety of real world constraints. First, an optimal analog transversal equalizer has an infinite number of taps, requiring infinite length. Additionally, as the electric signals propagate with the speed of light, manufacturable analog delay lines comprising lumped elements have not been able to emulate good delays with reasonable accuracy while meeting both economic and volume constraints. Moreover, transversal equalizers in communication channels generally require receipt of data at different transfer rates, providing an additional requirement that the delay characteristics of optimal transversal equalizers must be both variable and tightly controlled.
As a result of such real world constraints, the communications industry has implemented digital electronics in the construction of the equalizer as a substitute for the optimal analog transversal equalizer. In the construction of such a digital substitute device, as the equalizer tap spacing is the same as the bit rate (1/T), the output sampler from an optimal analog equalizer can be moved to the input of the digital transversal equalizer. Thus, the resulting digital arrangement typically includes an analog to digital converter (ADC) connected to the output of the sampler and clocked at the bit rate, along with a digital transversal equalizer comprising clocked shift registers as delay elements connected to the ADC. More particularly, the analog input signal is subjected to preliminary frequency-domain filtering (to reduce noise), and then sampled and converted to digital form by the ADC. The output of the ADC is provided to the digital transversal equalizer wherein digital signal processing takes place through the addition of successive sample values from cells of the shift register, weighted appropriately by digital multipliers to provide samples of the equalized signal. These samples are then used for data recovery and for self-synchronization. Such digital transversal equalizers are well known in the prior art and are also referred as "synchronous equalizers" (see the previously incorporated Gitlin et al. reference, p. 492). For additional discussion concerning digital equalization, see U.S. Pat. No. 4,146,840 entitled TECHNIQUE FOR OBTAINING SYMBOL TIMING FOR EQUALIZER WEIGHTS issued Mar. 27, 1979 to McRae et al.
Thus, the digital substitute for the optimum linear receiver actually comprises two different receiving filters connected in series: (1) a non-adaptive analog frequency-domain matched filter that facilitates an optimal signal/noise ratio at its output, and (2) a self-adaptive digital time-domain synchronous transversal equalizer that facilitates descrambling of ISI (through the self-adaptation of tap weight values).
The use of synchronous transversal equalizers in the time-domain filtering of input signals, however, requires that all frequency-domain filtering must be completed before sampling takes place in order to minimize the occurrence of aliasing of noise and unwanted signal components. As described in more detail in the previously incorporated Gitlin reference, such synchronous transversal equalizers do not have a controllable frequency response.
It will be recognized that specialists in communication systems have not been satisfied with the performance of this substitute digital synchronous equalizer, and in response have recently developed what is referred to as a "fractionally spaced equalizer" (FSE). The FSE filter operates in a similar fashion as the synchronous equalizer, but is much more complex. Generally, the initial sampling in an FSE filter is performed at two or three times the bit rate and the number of delay elements and taps in an FSE filter is correspondingly two or three times that of a synchronous equalizer. Additionally, the tap-weight adaptation circuitry also operates at two or three times the bit rate. The FSE filter also includes a relatively sophisticated digital "decimation" circuit, which reprocesses oversampled values and provides the output of the FSE filter which is finally sampled at the bit rate. The FSE filter overcomes many of the disadvantages of the digital substitute (at the expense of more complex circuitry) and, within bit-rate limits, can closely approximate the performance of the optimum analog transversal equalizer. As a result of the superior performance of the FSE filter, the telephone industry (which operates with signals in the KHz range) is now in the process of replacing synchronous equalizers with FSE filters. (For more discussion of the FSE filter, see the previously incorporated Gitlin et al. reference at pp. 495-500.)
However, as desirable the FSE filter may be for applications such as telephone communication systems, the increasingly higher transfer rates required in the disc drive industry (which presently operate with signals in the hundreds of MHz range), as well as the associated power and space constraints, provide limitations on the effectiveness of both the synchronous equalizer and the FSE filter in disc drive applications. The phenomenal growth in personal computers in the past decade led to unprecedented progress in all aspects of associated technology, including advancements in disc drive magnetic storage. At the present, recording (areal) densities in modern disc drives are approaching 1 Gbit/square inch, with drastically decreasing energy allocations per stored bit. At the same time, data transfer rates of greater than 200 Mbits/second have been achieved, and increasingly higher data transfer rates will continue to be demanded in the future.
Additionally, as disc drive form factors continue to decrease, it becomes increasingly necessary to integrate the disc drive functions into large scale integration (LSI) devices having small chip sizes and power consumption levels, capable of functioning with lower signal/noise ratios (as transfer rates increase) while maintaining or improving bit error rate levels (which are typically on the order of 10.sup.-12). The industry is also transitioning from peak detection to Partial Response, Maximum Likelihood (PRML) read channels in disc drives. The term "partial response" indicates that the response of a single bit transferred binary information is spread out to adjacent bit intervals; that is, only part of the bit response is inside of the corresponding nominal bit interval. In disc drives of this type, partial response signaling is utilized to control, rather than to suppress, intersymbol interference (ISI) and the effect of noise is minimized by the use of maximum likelihood detection of the magnetization of sequences of segments of the data track. To this end, signals corresponding to individual flux transitions are filtered to a signal which, in the absence of noise, would have a nominal form and the signals are then sampled at times determined in relation to this nominal form for maximum likelihood detection in which each bit of encoded data is recovered in the context of the sequence of bits that were written to the disc to limit the effect of noise.
PRML signal processing places stringent requirements on filtering of the signals induced in the read head prior to maximum likelihood detection. To achieve satisfactory results, the filtering must be performed in both the frequency-domain (to minimize the effects of noise) and in the time-domain (to obtain a particular waveform with known and controlled ISI, which can subsequently be descrambled in a maximum likelihood detector). While maximum likelihood detection limits the effect of noise and ISI, the variances between the nominal, ideal form to which the signal induced in the read head is to be filtered and the actually realized filtered signal constitute systematic noise which can generate errors in data recovery.
Another complicating factor in disc drive read channels is that, given constant rotational speed of the disc and relatively uniform areal recording density with respect to disc radius, the transfer rate of data from a data track at the outermost radius of the disc is generally about three times the transfer rate for a data track at the innermost radius. This requires a corresponding frequency scaling of the filtering system with respect to disc radius. Additionally, the form of the signal induced in a magnetic head (or AC sense circuitry used with magneto-resistive heads) varies not only from drive to drive, but also among heads within each drive. For economic reasons, obtaining sufficient yields in manufacturing of inexpensive disc drives necessitates the allowance of relatively loose tolerances in the variations of head-media components and the ability of the channel electronics to compensate for the effects of these variations. Thus, the filter in a disc drive must not only be capable of filtering the input signal to a reasonable approximation of a specific waveform, but must be able to do so adaptively, and at a low cost.
As provided hereinabove, prior art attempts to implement circuits for performing filtering in PRML channels in disc drives involved the use of preliminary frequency-domain filtering in combination with a synchronous digital transversal equalizer. In addition to the aforementioned problems associated with the inability of the synchronous equalizer to perform frequency-domain filtering, the use of such synchronous equalizers in the specific area of disc drives also have provided additional limitations. First, the digital signal processing and the inherent delay in the synchronous transversal equalizer requires at least several additional clock cycles, which introduces additional processing time in the timing circuitry (phased locked loop, or PLL) that provides the sampling clock for the ADC. As a result, time correction is delayed with respect to the times at which samples are taken. Such delay is usually termed "transportation lag" or "dead time" in control theory; see, for example, K. Ogata, "MODERN CONTROL ENGINEERING", Prentice-Hall, Englewood Cliff, 1970, pp. 346-350. This dead time adversely affects the stability of the PLL. Additionally, as transfer rates continue to increase, the digital signal processing must occur at very high speeds, requiring increasingly more complex circuitry with larger (silicon die) areal and power consumption requirements. These limitations provide an upper limit to the feasibility of the use of synchronous equalizers in disc drive PRML read channels.
The disadvantages of digital synchronous equalizers have led to attempts to replace the digital time-domain filtering with analog time-domain elements through the use of analog transversal equalizers. However, these attempts have been largely unsuccessful, primarily due to the constraints that led the industry to use digital electronics in the first place as a substitute for the optimum analog transversal equalizer. The problems associated with such prior art attempts to use analog transversal equalizers is illustrated by U.S. Pat. No. 5,325,130 entitled GHOST CANCELER, issued Jun. 28, 1994 to R. Miller, which describes a programmable analog transversal equalizer for a High Definition Digital Television (HDDT) application. Because of the absence of adequate electronic delay elements, the equalizer of Miller employs the use of an "exotic" electro-acoustic delay device in order to achieve reasonable delay performance. While such an approach may be feasible in an HDDT application, such an approach is not feasible in a disc drive read channel due to both size and cost constraints.